1. Technical Field
The present invention relates to, in a radio communication, a radio communication device in which loaded conditions of radio nodes which are present in communication partners and communication routes are detected so that an optimum communication partner and communication route are selected in accordance with the loaded conditions so detected, and a communication load calculation method for the same system.
2. Background Art
In a radio communication such as a wireless LAN, selecting a radio node with little load as a communication partner becomes not only important to realize a higher throughput and a low delay but also useful in making effective use of the resources of the whole communication network through load distribution.
Incidentally, in a wireless LAN specified under IEEE 802.11 illustrated in Non-Patent Document 1, with IEEE 802.11a, a communication speed is specified as a maximum transmission rate of 54 Mbps. However, this transmission rate is a rate for a physical layer, and hence, an actual communicable throughput is reduced by a CSMA/CA access control in an MAC (Media Access Control) layer or the like.
Here, the CSMA/CA access control which utilizes the DCF specified under IEEE 802.11 will briefly be described by reference to page 76 of the “9.2.5 DCF access procedure” and page 83 of the “9.2.8 ACK procedure” of Non-Patent Document 1.
FIG. 19 is a diagram which simply illustrates a communication time when data is transmitted using the access control called DCF (Distributed Coordination Function).
Firstly, before starting a communication, a transmission is waited for by a fixed length of time called DIFS (Distributed Interframe Space) and a random length of time called backoff. Next, a radio frame (described as DATA in the figure) which is actual transmission data is transmitted, and finally, in order to confirm a proper delivery of the radio frame to a target terminal, waiting for a length of time called SIFS (Short Interframe Space) and an ACK (Acknowledge) response from the target terminal is performed, and the reception of the ACK response completes the transmission of the radio frame.
In the frame transmission time illustrated in FIG. 19, it is only a portion of actual data transmission time that is to be changed in length of time by the size of data to be transmitted. In other words, irrespective of the size of data to be transmitted, the extra time of “transmission waiting time+delivery confirmation response time” is generated at all times. Consequently, in the event that short data of the order of 256 bytes are transmitted, compared with a case where long data of the order of 1500 bytes are transmitted, the ratio of “transmission waiting time+delivery confirmation time” to the one frame transmission time becomes large. Because of this, in the wireless LAN of IEEE 802.11, the shorter the frame length (or the packet length which results by removing the frame header from the frame) becomes, the lower the communication efficiency becomes.
Here, throughputs associated with two packet lengths will be calculated using an expression for calculating a throughput which is described on page 100 of Non-Patent Document 2. When the length of an IP packet is 1500 bytes, a throughput becomes about 31 Mbps, while when the length of an IP packet is 256 bytes, a throughput becomes about 10 Mbps, and it is seen from this that three times or more throughput difference is generated depending upon packet lengths.
Incidentally, in recent years, a telephone service using IP, that is, so-called IP phones have been in propagation, and due to its easiness in handling, there has emerged case where the IP phones are used in the wireless LANs. In the IP phones, conversations are implemented thereover by exchanging voice packets with a communication partner as communication data, and in voice packets, although depending upon compression systems, the packet length becomes shorter in general.
For example, the length of a code when coding is implemented by the PCM used under G.711 becomes 64 Kbps. In the event that a voice packet is transmitted by being packetized every 20 ms so that a large delay is not generated, the data volume of a voice part becomes about 160 bytes. Even when a header such as RTP, UDP, IP or the like is added to this voice part, the resulting data volume is on the order of about 200 bytes (and even when a frame header of the wireless LAN is also added thereto, the resulting data volume becomes on the order of about 240 bytes).
In addition, when using a voice encoding which has a higher compression rate as in G.729, the data volume of the voice part becomes smaller. When many packets such as voice having such a short packet length are transmitted using radio waves, there may be caused a situation in which the load is increased high even when a band which is used in an actual communication is not large. For example, assume that an actual communication band is 7 Mbps with the transmission rate being 54 Mbps in 802.11a. In addition, assume that a threshold which determines a high load is a load factor of 60% or more. In this situation, in case an average packet length in an actual communication is 1500 bytes, a throughput becomes about 31 Mbps, and because of this, the load factor becomes 23%, which is not a high load. When the average packet length is 256 bytes, however, since the throughput becomes about 10 Mbps, the load factor becomes about 70%, which is a high load.
Furthermore, in the case of the transmission control protocol or TCP, since it is a communication protocol which is redundant by such an extent that TCP ACK is returned by a communication partner, a communicable band becomes narrow compared with UDP. Because of this, a communication band which is determined to be highly loaded is much narrower than UDP.
On the other hand, it is described in Patent Document 1 and Patent Document 2 as related to load distribution, that load is determined based on a total data volume to be transmitted or bandwidth. Conventionally, in this way, the state of load has been grasped simply according to a band which is used currently without depending upon packet length.
In addition, as a technique in relation to QoS, it is described in Patent Document 3 that a band is guaranteed by totalizing traffics and changing access point of radio nodes in accordance with the totalized traffics between base stations. Conventionally, in this way, the QoS control has been implemented based on the traffic amounts without depending upon packet length.
[Patent Document 1] JP-A-2004-320274
[Patent Document 2] JP-T-2005-530378
[Patent Document 3] JP-A-2003-060663
[Non-Patent Document 1] ANSI/IEEE Std 802.11, 1999 Edition Part 11: Wireless LAN Medium Access Control (MAC) and PHYsical Layer (PHY) Specifications
[Non-Patent Document 2] “802.11 High-speed Wireless LAN Textbook” by IMPRESSNET BUSINESS COMPANY, on Mar. 29, 2003
However, as in the aforesaid wireless LAN, since a communicable band changes depending upon a packet length which is used in an actual communication, in a communication state where there are communicated many short packets, in the case of Patent Document 1 and Patent Document 2, there occurs a case where a highly loaded state is not determined although it exists in reality. In these circumstances, an accurate loaded state cannot be grasped by the calculation of a loaded state of a radio node based on simple communication volume, and the radio terminal cannot select accurately a radio AP which is little loaded.
In addition, the same problem is provided not only between a radio terminal and a radio AP but also between radio nodes which implement a multi-hop transfer as in a mesh network or ad hoc network. Namely, when a route including a radio node which is little loaded is selected as a transferring destination in multi-hop transfer, in the event that a load calculation method does not take packet length into consideration, an accurate loaded state cannot be calculated, and as a result, a route which is little loaded cannot be selected accurately.
Additionally, in the case of the radio AP which provides the band guarantee service, in the event that an average packet length of a currently used band is not taken into consideration as in Patent Document 3, or in the event that an average packet length on the band requesting side is not taken into consideration, it cannot be determined accurately whether or not the requested band can really be guaranteed. For example, in the event that a band of 12 Mbps is requested when communications are carried out under IEEE 802.11a whose transmission rate is 54 Mbps, according to the conventional method, it is determined that the band can be guaranteed in case there exists a radio node whose current band utilization is zero. However, in reality, the requested band guarantee depends on the average packet length on the band requesting side, and since in case the average packet length is 1500 bytes, the requested band can be guaranteed, but in case the average packet length is 256 bytes, a throughput that can be provided is 10 Mbps at maximum, as a result, the requested band cannot be guaranteed, and this means that the wrong determination has been made above.